公开(公告)号：US20100245111A1

公开(公告)日：2010-09-30

申请号：US12303824

申请日：2007-12-07

申请人： Tomas F. Gaensler ,  James A. Johanson ,  Peter Kroon ,  Ashish Parajuli ,  Richard Verney

发明人： Tomas F. Gaensler ,  James A. Johanson ,  Peter Kroon ,  Ashish Parajuli ,  Richard Verney

IPC分类号： G08C19/00

CPC分类号： H04M3/4285 ,  H04M1/72513 ,  H04M1/72558 ,  H04M2203/2011 ,  H04W4/16

摘要： In an exemplary embodiment, a wireless handset allows a user having a connection in an “on-hold” state to select one or more sources for play-out of media at a handset receiver while in the on-hold state, and then be signaled when the on-hold state is terminated. Such on-hold state might be indirectly detected, such as by detection of music-on-hold, or directly detected through on-hold notification. User selected media for play-out might be locally generated at the user's handset, or provided through a separate connection established between the wireless handset and the network.

摘要翻译： 在示例性实施例中，无线手机允许具有处于“保持”状态的连接的用户在处于保持状态的同时，在手持机接收机处选择一个或多个源以播放媒体，然后被发信号通知 当保持状态被终止时。 可以间接地检测这种保持状态，例如通过检测保持音乐，或通过保持通知直接检测。 用户播放的用户选择的媒体可以在用户的​​手机本地生成，或者通过在无线手机和网络之间建立的单独的连接提供。

公开(公告)号：US5732389A

公开(公告)日：1998-03-24

申请号：US482708

申请日：1995-06-07

申请人： Peter Kroon ,  Yair Shoham

发明人： Peter Kroon ,  Yair Shoham

IPC分类号： G10L19/08 ,  G10L11/06 ,  G10L19/00 ,  G10L19/04 ,  H03M7/30 ,  H04B14/04 ,  G10L9/14

CPC分类号： G10L25/93 ,  G10L2025/932

摘要： A CELP speech decoder includes a first portion comprising an adaptive codebook and a second portion comprising a fixed codebook. The CS-ACELP decoder generates a speech excitation signal selectively based on output signals from said first and second portions when said decoder fails to receive reliably at least a portion of a current frame of compressed speech information. The decoder does this by classifying the speech signal to be generated as periodic (voiced) or non-periodic (unvoiced) and then generating an excitation signal based on this classification. If the speech signal is classified as periodic, the excitation signal is generated based on the output signal from the first portion and not on the output signal from the second portion. If the speech signal is classified as non-periodic, the excitation signal is generated based on the output signal from said second portion and not on the output signal from said first portion.

摘要翻译： CELP语音解码器包括包括自适应码本的第一部分和包括固定码本的第二部分。 当所述解码器不可靠地接收到压缩语音信息的当前帧的至少一部分时，CS-ACELP解码器基于来自所述第一和第二部分的输出信号选择性地产生语音激励信号。 解码器通过将要生成的语音信号分类为周期性（有声）或非周期性（无声），然后基于该分类产生激励信号来实现。 如果语音信号被分类为周期性，则基于来自第一部分的输出信号而不是来自第二部分的输出信号产生激励信号。 如果语音信号被分类为非周期性，则基于来自所述第二部分的输出信号而不是来自所述第一部分的输出信号产生激励信号。

公开(公告)号：US5680506A

公开(公告)日：1997-10-21

申请号：US368059

申请日：1994-12-29

申请人： Peter Kroon ,  Suhas A. Pai ,  Frank Kao-Ping Soong

发明人： Peter Kroon ,  Suhas A. Pai ,  Frank Kao-Ping Soong

IPC分类号： G10L15/26 ,  G10L19/14 ,  G10L3/02 ,  G10L5/06 ,  G10L9/14

CPC分类号： G10L19/16 ,  G10L15/26 ,  G10L25/12 ,  G10L25/24

摘要： The present invention provides a novel method of analyzing speech signals in order to reduce the computational power required to perform both speech compression and voice recognition operations. Digital speech signals are provided to a speech analyzer which generates a linear predictive coded (LPC) speech analysis signal that is compatible for use in both the voice recognition circuit and the speech compression circuit. The speech analysis signal is then provided to the compression circuit, which further processes the signal into a form used by an encoder and then the encoder encodes the processed signal. The same speech analysis signal is also provided to a voice recognition circuit, which further processes the signal into a form used by a recognizer and then the recognizer performs recognition on the processed signal.

摘要翻译： 本发明提供了一种分析语音信号的新方法，以便降低执行语音压缩和语音识别操作所需的计算能力。 数字语音信号被提供给语音分析器，语音分析器产生与语音识别电路和语音压缩电路两者兼容的线性预测编码（LPC）语音分析信号。 然后将语音分析信号提供给压缩电路，该压缩电路进一步将信号处理为编码器使用的形式，然后编码器对经处理的信号进行编码。 语音分析信号也被提供给语音识别电路，语音识别电路进一步将信号处理成识别器使用的形式，然后识别器对处理过的信号进行识别。

公开(公告)号：US20080079733A1

公开(公告)日：2008-04-03

申请号：US11536177

申请日：2006-09-28

申请人： Richard Benson ,  Peter Kroon ,  Nigel Henry Wood

发明人： Richard Benson ,  Peter Kroon ,  Nigel Henry Wood

IPC分类号： G06F15/16

CPC分类号： H04N19/423 ,  H04N19/61

摘要： In a system comprising a plurality of processors and a memory shared by at least a subset of the processors, a method for processing video data includes the steps of: (a) a first one of the processors receiving a first video frame and storing the first video frame in the memory; (b) the first one of the processors receiving at least a second video frame, receipt of the second video frame initiating a release of the first video frame from the memory; (c) the first one of the processors sending the first and second video frames to a second one of the processors together for processing by the second one of the processors; (d) the second one of the processors generating an output video frame based at least on the first and second video frames; (e) storing the output video frame in the memory by overwriting an available memory location therein, the output video frame becoming a new first video frame; and (f) repeating steps (b) through (e) until all video frames to be processed have been received.

摘要翻译： 在包括多个处理器和由处理器的至少一个子集共享的存储器的系统中，用于处理视频数据的方法包括以下步骤：（a）处理器中的第一个处理器接收第一视频帧并存储第一 视频帧在内存中; （b）处理器中的第一个接收至少第二视频帧，第二视频帧的接收从存储器发起第一视频帧的释放; （c）处理器中的第一个处理器将第一和第二视频帧发送到处理器中的第二处理器，用于由处理器中的第二处理器进行处理; （d）所述第二处理器至少基于所述第一和第二视频帧产生输出视频帧; （e）通过重写其中的可用存储器位置将输出视频帧存储在存储器中，输出视频帧变为新的第一视频帧; 和（f）重复步骤（b）至（e），直到已经接收到要处理的所有视频帧。

公开(公告)号：US06920422B2

公开(公告)日：2005-07-19

申请号：US10039458

申请日：2001-11-07

申请人： Peter Kroon ,  Deepen Sinha

发明人： Peter Kroon ,  Deepen Sinha

IPC分类号： G10L19/02 ,  G10L19/00 ,  G10L19/14 ,  H03M7/30 ,  H04M11/00 ,  H04N7/24 ,  H04N7/26 ,  G10L19/08

CPC分类号： G10L19/24 ,  H03M7/30

摘要： An apparatus for providing at least first and second representations of an audio signal for use in a communications system is described. The apparatus comprises a first quantizer for quantizing at least a portion of the signal in accordance with a first multidimensional lattice to generate a first representation. The apparatus further comprises a second quantizer for quantizing at least a portion of the signal in accordance with a second, different multidimensional lattice to generate a second representation. In an illustrative embodiment, the first representation is a core representation containing core audio information. The second representation is an enhancement representation containing enhancement audio information. The core representation is necessary for recovering the audio signal with minimal acceptable quality. Audio quality is enhanced when the core representation, together with the enhancement representation, is used to recover the audio signal. A method for use in such an apparatus is also described.

摘要翻译： 描述了一种用于提供用于通信系统中的音频信号的至少第一和第二表示的装置。 该装置包括第一量化器，用于根据第一多维晶格量化至少一部分信号以产生第一表示。 该装置还包括第二量化器，用于根据第二不同的多维网格对信号的至少一部分进行量化以产生第二表示。 在说明性实施例中，第一表示是包含核心音频信息的核心表示。 第二表示是包含增强音频信息的增强表示。 以最小可接受的质量恢复音频信号需要核心表示。 当使用核心表示以及增强表示来恢复音频信号时，音频质量得到增强。 还描述了在这种装置中使用的方法。

公开(公告)号：US06728344B1

公开(公告)日：2004-04-27

申请号：US09356773

申请日：1999-07-16

申请人： Vasu Iyengar ,  Peter Kroon

发明人： Vasu Iyengar ,  Peter Kroon

IPC分类号： H04M164

CPC分类号： H04M1/6505

摘要： A telephone answering device including two separate coders, a first coder for encoding/decoding fixed voice prompts spoken by a single speaker, and a second coder for encoding/decoding incoming and outgoing voice messages spoken by multiple speakers. The first coder uses a first set of codebooks trained based on a first set of utterances spoken by a single speaker, while the second coder uses a second set of codebooks trained based on a second set of utterances spoken by multiple speakers. Because the first set of utterances is significantly smaller in size than the second set of utterances, and the range of pitch period is significantly smaller in size for the first set of utterances spoken by a single speaker in comparison to that of the second set of utterances spoken by multiple speakers, the size of the first set of codebooks is significantly reduced relative to the size of the second set of codebooks. As a result, the fixed voice prompt messages may be compressed at a lower bit rate with a relatively high quality of encoding, thereby optimizing the codebook and reducing the amount of necessary memory capacity for storing the encoded fixed voice prompts. The memory required for the encoded first voice prompts is so small that they can be stored in a low cost DSP ROM.

摘要翻译： 包括两个单独的编码器的电话应答设备，用于编码/解码由单个扬声器说出的固定语音提示的第一编码器，以及用于编码/解码多个扬声器所说出的进入和离开语音消息的第二编码器。 第一编码器使用基于由单个扬声器说出的第一组话语训练的第一组码本，而第二编码器使用基于由多个扬声器说出的第二组话语训练的第二组码本。 因为第一组话语的大小明显小于第二组话语，并且与第二组话语相比，单个讲话者所说的第一组话语的音调周期的范围显着地小得多 由多个扬声器说话，相对于第二组码本的大小，第一组码本的大小显着降低。 结果，可以以较高的编码质量以较低的比特率压缩固定的语音提示消息，从而优化码本并减少存储编码的固定语音提示所需的存储容量的量。 编码的第一语音提示所需的存储器非常小以至于可以将其存储在低成本的DSP ROM中。

公开(公告)号：US5848098A

公开(公告)日：1998-12-08

申请号：US679404

申请日：1996-07-09

申请人： James Y. Cheng ,  Shao-Ning Chou ,  Peter Kroon

发明人： James Y. Cheng ,  Shao-Ning Chou ,  Peter Kroon

IPC分类号： H04W88/18 ,  H04B1/38 ,  H04L5/16

CPC分类号： H04W88/181

摘要： A base station includes a digital signal processor, the base station operable to communicate with at least first and second wireless terminals using compressed digital signals modulated onto a radio frequency carrier, the base station further operable to communicate with an external network to facilitate a call between one or more wireless terminals and another party connected to the external network. The digital signal processor is operably connected to receive compressed signals from and provide compressed signals to the first and second wireless terminals, and further operably connected to communicate uncompressed signals with the external network. The digital signal processor is programmed to execute instructions to perform the following functions: monitor a first signal, the first signal received from the first wireless terminal; monitor a second signal, the second signal received from the second wireless terminal; monitor a network signal, the network signal received from the external network; select using predetermined criteria a priority signal, the priority signal comprising one of the first signal, the second signal and the network signal; and perform either a compression or decompression process on the priority signal and then provide the processed priority signal to at least one of the first wireless terminal, second wireless terminal and external network.

摘要翻译： 基站包括数字信号处理器，所述基站可操作以使用调制到无线电频率载波上的压缩数字信号与至少第一和第二无线终端进行通信，所述基站还可操作以与外部网络进行通信， 一个或多个无线终端和连接到外部网络的另一方。 数字信号处理器可操作地连接以从第一和第二无线终端接收压缩信号并向第一和第二无线终端提供压缩信号，并进一步可操作地连接以与外部网络传送未压缩信号。 数字信号处理器被编程为执行指令以执行以下功能：监视第一信号，从第一无线终端接收的第一信号; 监视从第二无线终端接收的第二信号的第二信号; 监控网络信号，从外部网络接收的网络信号; 选择使用预定标准的优先级信号，所述优先级信号包括第一信号，第二信号和网络信号之一; 对优先级信号执行压缩或解压缩处理，然后将经处理的优先级信号提供给第一无线终端，第二无线终端和外部网络中的至少一个。

公开(公告)号：US20140226560A1

公开(公告)日：2014-08-14

申请号：US13762408

申请日：2013-02-08

申请人： Jerome Parron ,  Peter Kroon

发明人： Jerome Parron ,  Peter Kroon

IPC分类号： H04W24/02

CPC分类号： H04W24/02 ,  H04L1/205 ,  H04L47/283 ,  H04L67/322

摘要： A radio communication device may be provided. The radio communication device may include: a receiver configured to receive data; a buffer configured to buffer a variable amount of the data; a reception condition determiner configured to determine a reception condition indicating a condition under which the receiver receives the data; and a buffer amount setter configured to set the amount of the data based on the determined reception condition.

摘要翻译： 可以提供无线电通信设备。 无线电通信设备可以包括：接收器，被配置为接收数据; 配置为缓冲可变数量的数据的缓冲器; 接收条件确定器，被配置为确定指示接收者接收数据的条件的接收条件; 以及缓冲量设定器，被配置为基于所确定的接收条件来设置数据量。

公开(公告)号：US4932061A

公开(公告)日：1990-06-05

申请号：US841906

申请日：1986-03-20

申请人： Peter Kroon ,  Edmond F. A. Deprettere ,  Robert J. Sluyter

发明人： Peter Kroon ,  Edmond F. A. Deprettere ,  Robert J. Sluyter

IPC分类号： G01L9/14 ,  G10L19/10

CPC分类号： G10L19/10

摘要： A multi-pulse excitation linear-predictive speech coder operates in accordance with an analysis-by-synthesis method for determining the excitation. The coder (10) comprises an LPC-analyzer (11), a multi-phase excitation generator (13), means (12, 14) for forming an error signal representative of the difference between an original speech signal (s(n)) and a synthetic speech signal (s(n)), a filter (15) for perceptually weighting the error signal and means (16) responsive to the weighted error signal (e(n)) for generating pulse parameters controlling the excitation generator (13) so as to minimize a predetermined measure of the weighted error signal. The LPC-parameters and the pulse parameters of the excitation signal (x(n)) are encoded for efficient storage or transmission. The bit capacity required for pulse position encoding of the excitation signal (x(n)) is considerably reduced by arranging the excitation generator (16) for an excitation signal (x(n)) which in each excitation interval (L) consists of a pulse pattern having a grid of a predetermined number (q) of equidstant pulses and by arranging the control means (16) for generating pulse parameters characterizing the grid position (k) relative to the beginning of the excitation interval (L) and the variable amplitudes (b.sub.k (j), 1.ltoreq.j.ltoreq.q) of the pulses of the grid.

摘要翻译： 多脉冲激励线性预测语音编码器根据用于确定激励的合成分析方法进行操作。 编码器（10）包括LPC分析器（11），多相励磁发生器（13），用于形成表示原始语音信号（s（n））之间的差的误差信号的装置（12,14） 以及合成语音信号（s（n）），用于感知地加权误差信号的滤波器（15）和响应于加权误差信号（e（n））的装置（16）），用于产生控制激励发生器 ），以便最小化加权误差信号的预定测量。 激励信号（x（n））的LPC参数和脉冲参数被编码用于有效的存储或传输。 激励信号（x（n））的激励信号（x（n））的激励信号（x（n））的脉冲位置编码所需的位容量大大降低，激励信号（x（n））在每个激励间隔（L） 具有预定数量（q）等距脉冲的网格的脉冲图案，并且通过布置用于产生表征网格位置（k）的相对于激励间隔（L）的开始的脉冲参数的控制装置（16）和可变幅度 （bk（j），1

公开(公告)号：US07693721B2

公开(公告)日：2010-04-06

申请号：US11953382

申请日：2007-12-10

申请人： Frank Baumgarte ,  Peter Kroon

发明人： Frank Baumgarte ,  Peter Kroon

IPC分类号： G06F15/00 ,  G10L11/00 ,  G10L19/00

CPC分类号： H04S1/007 ,  G10L15/22 ,  G10L19/008 ,  G10L19/0204 ,  G10L19/0212 ,  H04S3/002 ,  H04S3/02 ,  H04S5/00 ,  H04S5/005 ,  H04S2420/03

摘要： Part of the spectrum of two or more input signals is encoded using conventional coding techniques, while encoding the rest of the spectrum using binaural cue coding (BCC). In BCC coding, spectral components of the input signals are downmixed and BCC parameters (e.g., inter-channel level and/or time differences) are generated. In a stereo implementation, after converting the left and right channels to the frequency domain, pairs of left- and right-channel spectral components are downmixed to mono. The mono components are then converted back to the time domain, along with those left- and right-channel spectral components that were not downmixed, to form hybrid stereo signals, which can then be encoded using conventional coding techniques. For playback, the encoded bitstream is decoded using conventional decoding techniques. BCC synthesis techniques may then apply the BCC parameters to synthesize an auditory scene based on the mono components as well as the unmixed stereo components.

摘要翻译： 使用常规编码技术对两个或更多个输入信号的频谱的一部分进行编码，而使用双耳提示编码（BCC）编码其余的频谱。 在BCC编码中，输入信号的频谱分量被下混合，并且生成BCC参数（例如，信道间级别和/或时间差）。 在立体声实现中，在将左声道和右声道转换为频域之后，左声道和右声道的频谱分量对被混合成单声道。 然后将单声道分量转换回到时域，以及未被混合的左和右声道频谱分量，以形成混合立体声信号，然后可以使用常规编码技术进行编码。 为了播放，使用传统的解码技术解码编码比特流。 然后，BCC合成技术可以应用BCC参数以基于单声道组件以及未混合的立体声组件合成听觉场景。